[asterisk-dev] Re: SIPP Testing
Abdul Lateef Khan
abdulzu at hotmail.com
Wed Jun 7 22:42:24 MST 2006
Hello,
Can u post me xml file. i was trying with the following commond but it is
not working.
./sipp -sn uac -d 500 -s 123456 asteriskserver:1221 -l 300 -mp 1221 -i
asteriskserver:1221
Error:
Resolving asteriskserver...
media IP addr= asteriskserver
2006-06-08 01:41:35: Unable to bind RTP socket.
Could u please help to solve this issue?
>From: "Dan Austin" <Dan_Austin at Phoenix.com>
>Reply-To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
>To: "Asterisk Developers Mailing List" <asterisk-dev at lists.digium.com>
>Subject: RE: [asterisk-dev] Re: SIPP Testing
>Date: Wed, 7 Jun 2006 22:10:37 -0700
>
>The patch you seek is located in Mantis bugid 5374...
>
>Perhaps it would help. What is the hardware you are
>Testing against? If I read the chart correctly you could
>only setup about 3 calls per second which just doesn't
>seem right...
>
>-----Original Message-----
>From: asterisk-dev-bounces at lists.digium.com
>[mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Greg
>Boehnlein
>Sent: Wednesday, June 07, 2006 9:48 PM
>To: asterisk-dev at lists.digium.com
>Subject: [asterisk-dev] Re: SIPP Testing
>
>On Wed, 7 Jun 2006, Greg Boehnlein wrote:
>
> > Hello,
> > I am working on a bunch of issues, one of which is trying to
>track
> > down proof that using CONFIG_ZAPTEL_MMX w/ a Centos 4.3 2.6.9-34.0.1
> > kernel causes FPU errors w/ various codecs. This requires me to have a
>
> > high volume of SIP calls doing lots of Ulaw to G729 transcoding over a
>
> > period of time, using a modified g729 binary from Digium that logs
>when
> > the errors occur.
> > In doing this testing, I have setup a box w/ the latest SIPP
> > traffic generator (http://sipp.sourceforge.net) to make boatloads of
>120
> > second calls to a Music On Hold extension that simply plays Ulaw based
>
> > MOH to the caller.
>
>Just wanted to follow up on my post, archive it for the world and
>provide
>some insight into what the issue was.
>
>Asterisk's 1.2 RTP stack requires bidirectional traffic to actually send
>
>traffic back. There was a patch in Mantis a while back that did
>asynchronous RTP, but I was unable to find it. However, the fact that
>you
>need to receive an RTP packet to SEND an RTP packet is the reason why no
>
>RTP was originating from the Asterisk server.
>
>So.. here is what I did to get things working:
>
>1. Installed libpcap and libnet on my development box.
>2. Built SIPP w/ Pcap Play suppor using "make pcapplay"
>3. I grabbed a g729 RTP strean from a SNOM phone w/ TCPDUMP:
>
>tcpdump -T rtp -vvv dst 207.166.192.106 -w g729.pcap
>
>4. I then dumped the "uac_pcap" XML file and made some tweaks:
>
>./sipp -sd uac_pcap > uac_pcap.xml
>
>My edits consisted of modifying the following line:
>
><exec play_pcap="pcap/g711a.pcap"/>
>
>To
>
><exec play_pcap="pcap/g729.pcap"/>
>
>And, I changed:
>
>a=rtpmap:0 PCMU/8000
>
>To
>
>a=rtpmap:18 G729/8000
>
>5. I then executed the following:
>
>./sipp -sf uac-g729.xml -d 10000 -s 451 gw4.n2net.net -l 96 -mi
>207.166.192.254 -mp 5606 -i 207.166.192.254
>
>And presto.. LOTS of calls and LOTS of RTP traffic. Once you send one
>RTP
>packet to the Asterisk server, it responds back to the sipp RTP mirror
>port, and things just chug along after that..
>
>My results (not to promising)
>
>------------------------------ Scenario Screen -------- [1-4]: Change
>Screen --
> Call-rate(length) Port Total-time Total-calls Remote-host
>10.0(10000 ms)/1.000s 5060 877.47 s 2485
>207.166.192.184:5060(UDP)
>
> 0 new calls during 1.000 s period 2 ms scheduler resolution
> 2 concurrent calls (limit 96) Peak was 96 calls, after 9 s
> 1501 out-of-call msg (discarded)
> 1 open sockets
> 6390121 Total RTP pckts 236.576 last period RTP rate
>(kB/s)
>
> Messages Retrans Timeout
>Unexpected-Msg
> INVITE ----------> 2485 11980 2301
> 100 <---------- 144 0 45
> 180 <---------- 0 0 0
> 183 <---------- 0 0 0
> 200 <---------- E-RTD 139 674 0
> ACK ----------> 139 674
> [ NOP ]
> Pause [ 8000ms] 139 0
> Var Pause [ 10000ms] 139 0
> BYE ----------> 139 1209 130
> 200 <---------- 6 0 1
>
>------- Waiting for active calls to end. Press [Ctrl-c] to force exit.
>--------
>
>------------------------------ Scenario Screen -------- [1-4]: Change
>Screen --
> Call-rate(length) Port Total-time Total-calls Remote-host
>10.0(10000 ms)/1.000s 5060 878.45 s 2485
>207.166.192.184:5060(UDP)
>
> 0 new calls during 0.986 s period 1 ms scheduler resolution
> 0 concurrent calls (limit 96) Peak was 96 calls, after 9 s
> 1501 out-of-call msg (discarded)
> 1 open sockets
> 6397343 Total RTP pckts 234.380 last period RTP rate
>(kB/s)
>
> Messages Retrans Timeout
>Unexpected-Msg
> INVITE ----------> 2485 11980 2301
> 100 <---------- 144 0 45
> 180 <---------- 0 0 0
> 183 <---------- 0 0 0
> 200 <---------- E-RTD 139 674 0
> ACK ----------> 139 674
> [ NOP ]
> Pause [ 8000ms] 139 0
> Var Pause [ 10000ms] 139 0
> BYE ----------> 139 1209 132
> 200 <---------- 6 0 1
>
>------------------------------ Test Terminated
>--------------------------------
>
>
>----------------------------- Statistics Screen ------- [1-4]: Change
>Screen --
> Start Time | 2006-06-08 00:21:38
>
> Last Reset Time | 2006-06-08 00:36:15
>
> Current Time | 2006-06-08 00:36:16
>
>-------------------------+---------------------------+------------------
>--------
> Counter Name | Periodic value | Cumulative value
>-------------------------+---------------------------+------------------
>--------
> Elapsed Time | 00:00:00:986 | 00:14:38:484
>
> Call Rate | 0.000 cps | 2.829 cps
>
>-------------------------+---------------------------+------------------
>--------
> Incoming call created | 0 | 0
>
> OutGoing call created | 0 | 2485
>
> Total Call created | | 2485
>
> Current Call | 0 |
>
>-------------------------+---------------------------+------------------
>--------
> Successful call | 0 | 6
>
> Failed call | 2 | 2479
>
>-------------------------+---------------------------+------------------
>--------
> Response Time | 00:00:00:000 | 00:00:10:058
>
> Call Length | 00:00:50:108 | 00:00:32:841
>
>------------------------------ Test Terminated
>--------------------------------
>
>2006-06-08 00:36:16: Aborting call on UDP retransmission timeout for
>Call-ID '2429-17803 at 207.166.192.254'.
>sipp: There were more errors, enable -trace_err to log them.
>
>[root at dmx-64 sipp.cumulus.2006-03-20]#
>
>--
> Vice President of N2Net, a New Age Consulting Service, Inc. Company
> http://www.n2net.net Where everything clicks into place!
> KP-216-121-ST
>
>
>
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